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asterisk-opus package

asterisk-opus 13.7+20161113-3build1 source package in Ubuntu

Changelog

asterisk-opus (13.7+20161113-3build1) artful; urgency=medium

  * Rebuild against new asterisk 13.17.

 -- Gianfranco Costamagna <email address hidden>  Mon, 21 Aug 2017 13:23:32 +0200

Upload details

Uploaded by:
Gianfranco Costamagna
Uploaded to:
Artful
Original maintainer:
Debian VoIP Team
Architectures:
any
Section:
misc
Urgency:
Medium Urgency

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Series Pocket Published Component Section

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File Size SHA-256 Checksum
asterisk-opus_13.7+20161113.orig.tar.gz 23.3 KiB 869ccf8fc91aa5917cf69e434bab61547bd761dfcc55e889e432330ed4c2ba8c
asterisk-opus_13.7+20161113-3build1.debian.tar.xz 4.7 KiB d04b3c0321cb2f787b9bc6f1620f15e20acf14b7fcf2c60adcd54139bad2990f
asterisk-opus_13.7+20161113-3build1.dsc 2.0 KiB b2877b98ff8cbde0130ba1c4fa839ab7d904d853271e729f88496fe9af310956

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Binary packages built by this source

asterisk-opus: opus module for Asterisk

 Module for the Asterisk open source PBX which allows you to use the
 Opus audio codec.
 .
 Opus is the default audio codec in WebRTC. WebRTC is available in
 Asterisk via SIP over WebSockets (WSS). Nevertheless, Opus can be used
 for other transports (UDP, TCP, TLS) as well. Opus supersedes previous
 codecs like CELT and SiLK. Furthermore in favor of Opus, other
 open-source audio codecs are no longer developed, like Speex, iSAC,
 iLBC, and Siren. If you use your Asterisk as a back-to-back user agent
 (B2BUA) and you transcode between various audio codecs, one should
 enable Opus for future compatibility.
 .
 Opus is not only supported for pass-through but can be transcoded as
 well. This allows you to translate to/from other audio codecs like
 those for landline telephones (ISDN: G.711; DECT: G.726-32; and HD:
 G.722) or mobile phones (GSM, AMR, AMR-WB, 3GPP EVS).

asterisk-opus-dbgsym: debug symbols for asterisk-opus